Drop WebRTC audio data if OS has skipped frames.

media/base/audio_pull_fifo.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef MEDIA_BASE_AUDIO_PULL_FIFO_H_
 #define MEDIA_BASE_AUDIO_PULL_FIFO_H_
 
 #include "base/callback.h"
 #include "base/macros.h"
 #include "media/base/media_export.h"
@@ skipping 23 matching lines @@
 
   // Consumes |frames_to_consume| audio frames from the FIFO and copies
   // them to |destination|. If the FIFO does not have enough data, we ask
   // the producer to give us more data to fulfill the request using the
   // ReadCB implementation.
   void Consume(AudioBus* destination, int frames_to_consume);
 
   // Empties the FIFO without deallocating any memory.
   void Clear();
 
+  // Returns the size of the fifo in number of frames.
+  int SizeInFrames();
+
  private:
   // Attempt to fulfill the request using what is available in the FIFO.
   // Append new data to the |destination| starting at |write_pos|.
   int ReadFromFifo(AudioBus* destination, int frames_to_provide, int write_pos);
 
   // Source of data to the FIFO.
   const ReadCB read_cb_;
 
   // Temporary audio bus to hold the data from the producer.
   scoped_ptr<AudioBus> fifo_;
   int fifo_index_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioPullFifo);
 };
 
 }  // namespace media
 
 #endif  // MEDIA_BASE_AUDIO_PULL_FIFO_H_